The present invention relates to a device for digit rate reduction of PCM-signals.
PCM coding is well known. It comprises the steps of sampling analog signals at a so-called sampling frequency and quantizing, according to a logarithmic law, the samples at a rate of 8 binary elements per sample. Such a coding procedure has the drawback of requiring a high digit rate for achieving the quality required by the consumer, as it fails to exploit the statistical properties of the signal to be transmitted.
Digit rate reducing systems have already been described in the prior art. They allow a compression of the digit rate by utilizing, for the transmission of a sample of the signal, the knowledge acquired during the transmission of the preceding samples.
A system of this type has been described in an article by Mr. DIETRICH at the Zurich Seminar 1974. This system consists in an assembly of three cascade-connected stages. The first stage, called predictor stage, enables the PCM input to be replaced by a signal d.sub.n representing the difference between the PCM input signal and the predicted value of this sample as derived from the preceding sample. In the second stage, called automatic gain compression stage, the amplitude of the difference signal d.sub.n from the first stage is divided by an estimator of the mean power of the signal. A third stage, called switching quantizer, effects the optimum coding from the memorized quantizing characteristics which are best adapted to the conditional distributions of probability of the signal from the preceding stage and delivers at the output a digital signal reduced redundancy formed by words of fixed length, namely 4 binary elements per sample. A dual device at reception enables the PCM frame to be recovered and the message to be decoded.
However, such a system, which is satisfactory for the speech signals belonging to the frequency band 0,--3--3--4 KHz sampled at 8 KHz, is not suitable for the other signals which may be transmitted within the frequency band such as data signals, harmonic telegraphy signals or multifrequency signalling signals which are of different statistics. In fact, it is known that for speech signals the spectrum presents a maximum in the vicinity of 700 Hz but for data signals, this spectrum presents a maximum in the vicinity of 2 kHz. Thus, the predictor stage of Mr. DIETRICH which is optimized for the spectrum of the speech signals will no longer be adapted to the other types of signals.
In fact, the predictor stage usually consists of a filter disposed on a negative feedback loop and delivers a sample x.sub.pn predicted from coefficients C (i) derived from the autocorrelation function of the signal to be processed. However, prediction will be optimum only for a given type of signal and under the provision that this signal is stationary. Now, the speech signal is stationary only for a short time (about 10 ms), so that the coefficients of the predictor should be modified one hundred times per second if an always optimum prediction were desired. In addition, a fixed predictor optimized for a speech signal will provide a poor prediction for a data signal.
Other attempts have been made to achieve digital reduction with devices suitable for speech signals as well as data signals. Such devices generally comprise a fixed predictor the coefficients of which are determined as an acceptable compromise both for data and for speech.
However, the quality of transmission achieved with such systems is insufficient to allow these systems to be introduced into commercial telecommunication networks.